New answers tagged audio
1
Your set-up seems quite weird to me. Channel 1 is panned to the right, while channel 2/3 is also to the right. Shouldn't it be panned to the left, so that you can record each signal in a different track?
Also, you seem to connect both Left and Right output channels to the AutoHybrid input. This way, the person you are talking to would also hear his own ...
2
If you have music in midi format it is straightforward - just replace samples with your chip tune samples. Unfortunately there is no simple way to do this at raw wav level.
5
Your best bet will be to run your source material through a bitcrusher and a sample-rate reducer. However, that will just make it sound like the kind of low-quality samples that could be played back on old machines. Actual chiptune music tends to involve very careful use of low-resolution waveforms and FM synthesis.
5
The classic "tone stack" or three band EQ found on many guitar and bass amps is not usually implemented as three separate second order filters, but rather one third order filter with three parameters. As a result, the various "bands" aren't shaped and defined in the way you'd like. In particular, there is no "flat" setting and the range around 1kHz is ...
1
The feature you're looking for is called Audio Ducking and can be achieved by applying a side-chain compressor to your background track or by key framing your background audio volume to lower when the other track is playing.
A side-chain compressor monitors the amplitude of signal B and applies compression based on that to signal A. So if signal B will have ...
2
There is no such tool that I'm aware of and I'm not even sure what exactly the output of such a tool would look like. What is available, however, is the ability to use the integration between Adobe applications to work on the audio track in Premiere directly in Audition so that you shouldn't have to manage it yourself like this.
In fact, I believe it may ...
3
Actually, you're mostly doing it right, that's how many of us chiptune artists learn to do it: by messing around. Another thing you may want to try is using a tracker like Famitracker (or even a sample-based tracker like Milkytracker), and just creating sound effects with notes. Then add tracker effects to the notes.
2
I'm surprised nobody else has mentioned this, but the unit in the digital domain is dBFS, where FS stands for Full Scale. 0 dBFS represents the maximum digital signal level. A signal would clip in the digital domain above 0 dBFS. Note, however that just restraining a signal to just below 0 dBFS might not be enough to avoid clipping: Depending on the method ...
0
I believe that OSC (Open Sound Control) may help you in your problem. But I guess there won't be a quick solution to what you want. The fastest thing you could do is to buy some cables and make the appropriate connections (Presuming your devices are in an acceptable distance).
Such connectivity between two OS's is usually overkill, you need to ...
5
Once a signal is digitized it is treated as a number (as you quite rightly point out) and for 16bits the range of numbers are -32768 to +32767. The numbers are created by an analogue to digital converter.
The analogue to digital converter (ADC) will have a maximum input range from -X volts to +X volts (i.e. real signals that you could measure inside your ...
2
There is no direct or absolute correspondence between 'original' amplitude and sample value. The concept of dB itself is one of relative amplitude -- unless you specify a reference value, as in dBa or dBm, all you know about a decibel is that it represents a (logarithmic) ratio. There is no 'zero dB' as such.
In asking about original amplitude do you mean ...
0
The first thing to realize is that dB is a logarithmic scale. For every 10dB change, the signal level changes by an order of magnitude. 0db means the signal is unaltered when sampling or "full power" when dealing with a recording. -infinity is 0 signal strength and the amplitude of the signal can be anywhere in between.
To get a +dB level, ...
2
The M-Audio KeyStudio is a MIDI controller keyboard - it doesn't make any sound, rather, it sends MIDI messages to instruct something else to make sound. Since you're running it into FL Studio, I presume you intend to control some software instrument, so you will need to load that instrument, route the MIDI messages to it, and enable monitoring. All of that ...
0
I feel I should mention DropBox, GoogleDrive (with the new(?) desktop sync app) and Gobbler.
DropBox is my collaborative/online storage app of choice, but the other 2 have more free/initial storage space.
Gobbler is more of a backup and transfer service, but is optimised for media projects. Hopefully soon they will add a collaborative feature like ...
3
That appears to be a chassis common ground.
Update: After re-reading your question, I'm confused by what you mean by "which jacks are shielded?" The circuit diagram says nothing about whether the jacks are shielded or not. It does say that the grounds are connected to chassis ground. Most mixers have a metal body that acts as a shield for the entire device, ...
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